In previous years, Voice over internet protocol (VoIP) has become an important application and is expected to carry more and more voice traffic over TCP/IP networks.
In such Internet protocol (IP)-based voice communications systems, typically a voice waveform of a user is sliced in time, compressed by a voice coder, packetized, and transmitted to other users. Due to the inherent nature of IP networks and real-time constraint of human voice communications, it is common to lose voice packets during transmission or that late voice packets are discarded even if they are received, resulting in degraded voice quality. Mobile and WIFI networks usually make the situation worse in many cases. Thus, accurate real-time monitoring of voice quality is an advantagous feature for analysis, management, and optimization of voice communication systems.
A typical voice quality monitoring system adopts a scheme that analyzes packet loss information, such as packet loss rate and loss patterns (e.g., if the losses are random or of a bursty nature), as it provides a simple and computationally inexpensive way to estimate voice quality. This scheme is known as a modified E-model. However, these systems may suffer from low accuracy in estimating voice quality.
Many IP-based voice communications systems employ forward error correction (FEC) for recreating as many of the lost voice packets as possible. FEC provides the possibility of such recreation by adding redundant data streams, which are transmitted along with the voice packets. This may influence the voice quality as perceived by a user, since some of the lost voice packets may be recreated.
Thus, it is desirable to have a voice quality monitoring system utilizing both packet loss information and data about FEC.
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